similar to: Dialplan - intercoms

Hi, Does anybody know of a VoIP compatible doorbell or door intercom unit? I've contemplated buying a cheap SIP phone, ripping it apart, and putting it inside an IP66 sealed unit... It would need: - At least one speed-dial key, or some way to make every button dial the same extension number - PoE (power over ethernet), so I can power it off the central switch - cheap enough to rip apart

Hello I assume I'm not the first one to think about this: Is it possible to connect an intercom and/or door bell to Asterisk, so that I can get an e-mail that someone rang my place while I was out? Even better: If used for a doctor's office, it'd be cool if patients could type their Social Security Number on a keypad, which would open the door and notify Asterisk which would then

This is not specifically Asterisk-related, but I think that sometimes a "pre-emptive" clue is a good thing. Viking Electronics (http://www.vikingelectronics.com/) has some neat widgets that attach to phone lines, which I'm sure many of the people on this list would find at least somewhat useful in conjunction with their Asterisk systems. Many of the widgets can be replaced with

Asterisk as a phone system makes perfect sense in a condo. You can get all the DID's you want and eliminate costs for the owners. You can offer standard FXO for people who don't care and IP sets for people who want to "upgrade" to feature sets. Your door openner is a piece of cake. 1. Create an option in your dialplan only in the "from-access-door" context that reads

Hi Everyone, Looking to replace a condo intercom system. Apparently the current one taps into the lines and dials phone numbers but needs to be changed as it's faulty. I will probably still use the same analogue dialing and back it up with a VoIP line and use the current cabling that is in place. But as for as the door opening function goes, I am not sure how to interface and how open these

Very friggen cool, that you very much for the information it looks like it will do the job nicely. What did you use in your extensions list to activate the relay? James > -----Original Message----- > From: asterisk-users-bounces@lists.digium.com > [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Jay Milk > Sent: Sunday, 20 February 2005 6:24 PM > To: 'Asterisk

I wasn't thinking of using the conference system as the basis. I was thinking more along the lines of: 1) Setup a second extension on the Cisco phone named "INTERCOM" enabled for auto-answer 2) Create a call group on asterisk to dial that "INTERCOM" extension on every phone that will participate 3) Add a feature code that would dial the intercom extension and connect

Hi, Anyone know anything about Asterisk's support for door phones? Receiving the call from the door intercom system, opening the door, etc? Any hardware recommendations? I understand that the equipment we have now is Panasonic proprietary and came with the currently deployed Panasonic TD12-32 pbx. We intend to deploy Asterisk in a 72 extensions + 16 trunks in a while, so any info will be

Environment: Asterisk 1.2.14, FreePBX 2.2.0, Aastra 480i IP telephones firmware version 1.4.1.1077. Problem: Intercom feature: the dialed phone does not play the splash tone when auto-answering an intercom call. Otherwise, intercom works perfectly. Questions: What is the extensions.conf syntax to trigger a splash tone in Asterisk 1.2.14 (from the documentation and posts I've found, it has

A friend pointed out this url http://www.cisco.com/univercd/cc/td/doc/pcat/clmn32.htm where it lists intercom/auto-answer as being a feature in Cisco Call Manager (which as I understand it, uses SIP predominately for handsets). I've come across comment somewhere that intercom isn't supported in the SIP spec. Does anyone know if the apparent capability of Intercom being available in SIP

I'm having hard times with paging intercom Heres my dialplan exten => 777,1,Goto(intercom,777,1) [intercom] exten => 777,1,SIPAddHeader(Call-Info: <sip:192.168.16.105>\;answer-after=0) exten => 777,2,Page(Local/308 at page& Local/309 at page& Local/310 at page) [page] ; Paging context exten => _X.,1,Macro(page,SIP/${EXTEN}) [macro-page] ;

There was a thread a few months ago that tossed around some ideas for using a cisco phone for intercom or paging. I don't have any ip phones, and wondered if anyone had any luck getting intercom or paging to work on the cisco units. Do any of the (cheaper) ip phones have a way to support intercom or paging? I presume that it's not part of the SIP or IAX protocols. Chris.

Hi there, I'm new to this list, so sorry if this has already been asked before [hopefully this is the correct list too!]. I'm in the process of building my asterisk system. I've set up a CentOS machine, together with Cisco 7912 phone over SIP. I can make and receive calls from outside over my TDM400 card [2 FXO ports], everything works fine. My next project is to integrate or rebuild

Hi All... I'm trying to get the intercom feature working on some snom 190 phones but having no luck... As you can see from the SIP trace below (from the called phone), intercom=true is being appended to the To: header as per requirements. I've email'd snom a few days ago but have yet to get a response. On my 190s, im running snom190-SIP 3.57v. I am pulling the config for the

I am trying to get one of the function keys on the Snom 200 working as an intercom. However, I can't get the other Snom 200 phone to auto-answer. I found some posts in the archives from Christian that talk about intercom=true and also the Call-Info header. However, I can't get either one to work. I have tried firmware 2.04g,2.04h,2.05f,3.33 and none work. I am using chan_sip2z.c and

Today I released Enhance Me 1.004 for AMP 1.10 and Asterisk @ Home 2.2. These utilities allow for speed dialing, a revised version which uses AMP to store speed dial numbers and NAMES so that only operators of the web interface can add and delete speed dials. Data is stored in mysql instead of Asterisks' DB. This stops the waste of the 300 series extension numbers. Instructions on

Here is the link.. http://sourceforge.net/projects/enhanceme _____ From: Paul [mailto:paul@siliconvp.com] Sent: Wednesday, January 11, 2006 4:48 PM To: 'asterisk-users@lists.digium.com' Subject: Enchance Me 1.004 Released! Today I released Enhance Me 1.004 for AMP 1.10 and Asterisk @ Home 2.2. These utilities allow for speed dialing, a revised version which

Hey, Hi, All, We have a few dozen of the Kirk (ie: Polycom bought this European brand) out there & most all work very well & work very well with most all versions of Asterisk. But we have been tripped-up by one combination of firmware & version & configuration variables. We are running Asterisk 1.4.23.1 (TrixBox CE). We are running latest stable firmware on the handsets. Most

Greetings- As many of your are Polycom "experienced", I was hoping some kind soul could provide direction on a specific issue. On a system running Asterisk 11.11.0 (and latest FreePBX), I'm finding an instance where, using intercom/paging functionality of FreePBX, I need to override an end user's 'Do Not Disturb' selection on the handset. By default, DND simply

Hi everyone, I am looking for advice about the design of a SIP-based intercom. I count on your help, as my current attempts are not fruitful (yet). This will be a pretty long message, so here's my fundamental question: Is there a way to interpret DTMF tones sent by the calee (not the caller) while a voice call is in progress? Here's the desired scenario: - there is a box with